[asterisk-users] sip.conf templates and realtime

[Charles R. Wadsworth]

I currently have my phones setup in the sip.conf file.  I use templates
to describe the specific settings to each phone type.
For instance in sip.conf, I have:

[generic_phone](!)
...
...

[polycom501](!,generic_phone)
...
...

[grandstream](!,generic_phone)
...
...

;;;;;begin subscribers

[200](polycom501)
...
...

[201](grandstream)
...
...

I am using asterisk 1.4.21.2

I would like to move my sip users to realtime, so my questions are:

1)  Can I continue to use the templates from sip.conf and the template
settings get passed to realtime and if so, how?

In the comments in the sip.conf file where it shows the "User config
options" ant "Peer configuration", on the peer side it shows a
"template" field, which seems to indicate to me that this can be done.

2)  If this is not the purpose of the "template" field, what is it's
purpose?  I can not seem to find it documented anywhere.


Note:  I do not have any problems getting realtime to work, as long as I
put every field that is needed (or required) in each record, but I think
life would be easier if I could leave my templates (that rarely change)
in the sip.conf file and put the bare necessities in realtime (users
that change all the time).


Thanks,
Charles Wadsworth